WO1999046884A1 - Procede et appareil de mesure de la degradation du signal - Google Patents

Procede et appareil de mesure de la degradation du signal Download PDF

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Publication number
WO1999046884A1
WO1999046884A1 PCT/GB1999/000731 GB9900731W WO9946884A1 WO 1999046884 A1 WO1999046884 A1 WO 1999046884A1 GB 9900731 W GB9900731 W GB 9900731W WO 9946884 A1 WO9946884 A1 WO 9946884A1
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WO
WIPO (PCT)
Prior art keywords
error
measure
significance
data stream
classes
Prior art date
Legal status (The legal status is an assumption and is not a legal conclusion. Google has not performed a legal analysis and makes no representation as to the accuracy of the status listed.)
Ceased
Application number
PCT/GB1999/000731
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English (en)
Inventor
Paul Alexander Barrett
Current Assignee (The listed assignees may be inaccurate. Google has not performed a legal analysis and makes no representation or warranty as to the accuracy of the list.)
British Telecommunications PLC
Original Assignee
British Telecommunications PLC
Priority date (The priority date is an assumption and is not a legal conclusion. Google has not performed a legal analysis and makes no representation as to the accuracy of the date listed.)
Filing date
Publication date
Priority claimed from GBGB9805330.9A external-priority patent/GB9805330D0/en
Application filed by British Telecommunications PLC filed Critical British Telecommunications PLC
Priority to EP99907765A priority Critical patent/EP1062758A1/fr
Priority to US09/622,221 priority patent/US6714896B1/en
Publication of WO1999046884A1 publication Critical patent/WO1999046884A1/fr
Anticipated expiration legal-status Critical
Ceased legal-status Critical Current

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Classifications

    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
    • H04L1/00—Arrangements for detecting or preventing errors in the information received
    • H04L1/0001—Systems modifying transmission characteristics according to link quality, e.g. power backoff
    • H04L1/0014—Systems modifying transmission characteristics according to link quality, e.g. power backoff by adapting the source coding
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
    • H04L1/00—Arrangements for detecting or preventing errors in the information received
    • H04L1/0001—Systems modifying transmission characteristics according to link quality, e.g. power backoff
    • H04L1/0009—Systems modifying transmission characteristics according to link quality, e.g. power backoff by adapting the channel coding
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
    • H04L1/00—Arrangements for detecting or preventing errors in the information received
    • H04L1/0001—Systems modifying transmission characteristics according to link quality, e.g. power backoff
    • H04L1/0015—Systems modifying transmission characteristics according to link quality, e.g. power backoff characterised by the adaptation strategy
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
    • H04L1/00—Arrangements for detecting or preventing errors in the information received
    • H04L1/004—Arrangements for detecting or preventing errors in the information received by using forward error control
    • H04L1/0045—Arrangements at the receiver end
    • H04L1/0054—Maximum-likelihood or sequential decoding, e.g. Viterbi, Fano, ZJ algorithms
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
    • H04L1/00—Arrangements for detecting or preventing errors in the information received
    • H04L1/004—Arrangements for detecting or preventing errors in the information received by using forward error control
    • H04L1/0056—Systems characterized by the type of code used
    • H04L1/007—Unequal error protection
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
    • H04L1/00—Arrangements for detecting or preventing errors in the information received
    • H04L1/20—Arrangements for detecting or preventing errors in the information received using signal quality detector
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
    • H04L25/00—Baseband systems
    • H04L25/02—Details ; arrangements for supplying electrical power along data transmission lines
    • H04L25/03—Shaping networks in transmitter or receiver, e.g. adaptive shaping networks
    • H04L25/03006—Arrangements for removing intersymbol interference
    • H04L25/03178—Arrangements involving sequence estimation techniques

Definitions

  • the present invention relates to methods of, and apparatus for, measuring the degradation of a data stream. It has particular utility in relation to providing a measure for use in selecting a speech coding rate for use over a radio communication link between a mobile telephone and a fixed network.
  • degradation of a data stream has been estimated on the basis of a measurable parameter of the channel over which the data stream has been sent.
  • the degradation of coded speech carried over a radio channel has hitherto been estimated on the basis of a measured radio channel parameter such as the Carrier to Interferer (C/l) ratio or the Bit Error Rate.
  • C/l Carrier to Interferer
  • Bit Error Rate the Carrier to Interferer
  • European patent application no. 0 798 888 discloses a speech encoder/decoder for use in a mobile telephone.
  • the encoder divides the incoming speech into frames and derives a set of speech parameters for each frame.
  • the decoder is operable to receive decoded speech parameters for a frame and to determine whether they contain errors that are so significant that they are unsuitable for generating synthesised speech.
  • the encoder arranges the bits it outputs according to their importance to the quality of the decoded speech. Each bit is associated with an importance class which is in turn associated with a importance value.
  • a checksum is applied to each class of bits at the encoder.
  • the decoder analyses the checksum and thereby indicates whether the associated class contains any errors.
  • the importance values for those classes which contain errors are then added together and their sum is compared to a threshold. If the sum exceeds the threshold then the decoded parameters are deemed unsuitable for generating synthesised speech.
  • EP 0 798 888 discloses a measure of the degradation of one section of a data stream. When measuring the degradation of a data stream comprising a plurality of sections, it is not obvious that the measure for one section is any better than the conventional measurements of Bit Error Rate.
  • a first aspect of the present invention provides a method of measuring the degradation caused by the influence of errors on a data stream comprising a plurality of sections, said method comprising the steps of: retrieving stored error significance values associated with respective error significance classes of one or more digits of a section of said data stream; determining error likelihood measures for respective error likelihood classes of one or more digits in said section; calculating a degradation measure on the basis of the sum over said section of said error likelihood measure for each digit weighted by the error significance value associated with that digit.
  • Such a degradation measure i.e. one based on a sum over a section of an error likelihood estimate for each digit weighted by an error significance value for each digit
  • a Bit Error Rate or the like is a better measure of the degradation of a data stream than a Bit Error Rate or the like.
  • many coding schemes are designed to give an inverse correlation between the likelihood of a received digit being in error and the significance of an error in that digit. Where there is such a correlation, the coding scheme will operate to reduce the impact of an increase in Bit Error Rate.
  • the measure according to the present invention takes the effect of such coding schemes into account.
  • said digits will be bits (binary digits).
  • the invention might also be applied to other forms of coding such as quaternary or octal coding.
  • said error significance classes comprise bits at predetermined positions in the sections, and said error significance values retrieving step involves determining the class to which a bit belongs on the basis of the position of said bit within the section.
  • Some channel encoding/decoding algorithms provide a measure indicative of the likelihood of error in each decoded bit.
  • said error likelihood measures are provided on the basis of an error likelihood measure provided for each bit by a channel decoding method. This has the advantage that the calculation of the degradation measure can be implemented without increasing the bit-rate of the transmitted signal.
  • one application of the present invention is in the selection of a coding scheme to be used over a communication link between a sender which is operable to encode a signal using a selected one of a number of possible coding methods and a receiver. Therefore, according to a second aspect of the present invention, there is provided a method of selecting a signal encoding method for use over a communications link, said method comprising: carrying out a method according to the first aspect of the present invention to determine said degradation measure on the basis of a received portion of said data stream; and selecting one of a plurality of encoding methods on the basis of said degradation measure.
  • a method of measuring the degradation of a section of a data stream owing to the influence of errors thereon comprising the steps of: retrieving stored error significance values associated with respective error significance classes of one or more digits of said section; said method being characterised by: determining error likelihood estimates for respective error likelihood classes of one or more digits in said section; calculating a measure of said degradation on the basis of the sum over said digits of said error likelihood estimate for each digit weighted by the error significance value associated with that digit.
  • Figure 1 is a schematic illustration of part of a network providing mobile telephony
  • Figure 2 is a schematic illustration of components of a channel decoder used in the network.
  • Figure 3 is a flow chart illustrating the operation of the receiver which contains the channel decoder.
  • a first embodiment (a mobile radio speech transmission system) is schematically illustrated in Figure 1 .
  • a mobile telephone 10 includes a microphone 1 2 which receives a user's voice and samples it at 1 25 microsecond intervals to provide a binary data stream.
  • This data stream is then passed to an speech encoder 14 which operates on one block of samples at a time (the blocks of samples are known as a 'frames' in the art) to provide a compressed data stream representing the user's speech.
  • the compressed data stream passes to a channel encoder 16 which further adds error detection and correction bits to the compressed data stream to provide a transmission data stream (in practice, the mobile phone 10 will also encrypt the speech data at some stage in the encoding process) .
  • the transmission data stream is then used to modulate a radio carrier signal which is transmitted from the antenna 1 8 of the mobile phone 10 across a radio link 20 to a base station 30.
  • the mobile telephone 10 is operable to use three different speech/channel coding methods. In all three cases, the blocks of samples input to the speech encoder 14 represent 20ms of the user's voice.
  • a first speech/channel coding method involves the phone 10 operating in accordance with the Global System for Mobile communications (GSM) full-rate standard.
  • GSM Global System for Mobile communications
  • each input speech frame is processed by the speech encoder 14 to give a set of 260 bits (i.e. the speech is coded at 1 3kbits _1 ).
  • 1 82 of the 260 bits are known as class 1 bits
  • the remaining 78 bits are known as class 2 bits.
  • the set of class 1 bits comprises 50 class 1 A bits and 1 32 class 1 B bits.
  • the class 1 A bits are deemed especially important to speech quality.
  • the class 1 B bits are deemed to be less important than the class 1 A bits, but more important than the class 2 bits.
  • the channel encoder 1 6 applies a 3-bit Cyclic Redundancy Checksum to the class 1 A bits before adding four tail bits and performing convolution coding. These processes add 1 96 error detection and correction bits to the class 1 bits. In contrast, no further processing of the class 2 bits takes place in the channel encoder 1 6. This increases the data rate of the transmission data stream sent across the radio link 20 to 22.8kbits "1 .
  • a second coding method involves the speech encoder 14 producing speech coded at l Okbits "1 and the channel encoder 1 6 adding 1 2.8kbits '1 .
  • a third speech coding method involves the speech encoder 14 producing speech coded at 8kbits _1 and the channel encoder 1 6 adding 14-8kbits " ⁇
  • the selection of the speech coding method to be adopted by the mobile phone 10 in encoded the user's speech for transmission across the radio link 20 is made at the base station 30. Once the base station 30 determines the speech coding method to be adopted it sends a signal across the radio link 20 which controls the mobile phone 10 to encode speech using the selected method.
  • Figure 2 shows two components of a channel decoder 40 which forms part of the base station 30.
  • the radio carrier signal is demodulated at the base station 30 to recreate as accurately as possible the transmission data stream.
  • the recreated transmission data stream will normally differ from the transmitted data stream owing to errors being introduced in transmission across the link 20.
  • the soft output Viterbi equaliser 42 receives the recreated transmission data stream and, in effect, adjusts the data stream to compensate for measurable characteristics of the radio link 20.
  • the operation of the equaliser and the training sequences added to the transmitted data stream in order to control its operation will be understood by those skilled in the art.
  • the Viterbi equaliser 42 outputs both the estimated true value of each of the bits within the recreated transmission data stream and a value which indicates the probability that the bit is in error.
  • the equaliser operates in accordance with the well known Soft Output Viterbi Algorithm (SOVA) described in J. Hagenauer and P. Hoeher, "A Viterbi Algorithm with Soft-Decision Outputs and its Applications", Proc. GLOBECOM, pp1 680 - 1 686 ( 1 989), which is incorporated herein by reference.
  • SOVA Soft Output Viterbi Algorithm
  • sd, and p are the log likelihood value and probability of error for the f h bit of the transmission data stream respectively.
  • the log likelihood values for the class 2 bits are passed to a speech coding method selection unit 50.
  • the log likelihood values for the 378 bits generated from the 1 82 class 1 bits are passed to a soft output Viterbi decoder 44.
  • the Viterbi decoder 44 also uses the SOVA algorithm and regenerates the 1 82 class 1 bits together with a log likelihood value for each class 1 bit indicating the probability that the bit is in error. It will be realised that the log likelihood values associated with the class 1 bits will normally be higher than those associated with class 2 bits.
  • step 60 the log likelihood values for the class 1 bits are obtained. These values are then processed and averaged (step 62) to give a mean probability of error value for the class. This is done in accordance with the following equation:
  • sd i k is the log likelihood value associated with the bit of class k
  • L k is the number of bits in class k
  • p k is an average probability of an error in any given bit in class k (k being set to 1 in step 62, L 1 therefore being 1 82).
  • a process similar to that carried out in steps 60 and 62 is then carried out on the using the class 2 log likelihood values in steps 64 and 66.
  • Error significance values representing the significance of errors in the two classes are then obtained from memory in the speech coding and selection unit 50 (step 68).
  • One method by which these values (which are constant over time) might be arrived at is described later.
  • a sum over the classes of the averaged probability of error for the class weighted by the error significance values associated with the class is then calculated (step 70). This calculation may be represented by the equation:
  • E(SNR) represents the log weighted sum which is, in effect, an expected value of the signal to noise ratio of the recreated transmission data stream.
  • ⁇ is the length of the averaging filter in frames
  • E j (SNR) is the estimated expectation of the S ⁇ R calculated according to Equation (4) for frame /. For 20ms frames a good value of ⁇ / is 10, which is equivalent to a measurement window of t - 200ms.
  • the degradation measure thus calculated is then used to select the best channel coding method (step 76) as follows. For each coding method, two thresholds are defined and the following rules used to select the channel coding method at time t based on the value of M calculated over the most recent ten frames:
  • Th2 ⁇ Th l If current method is B: if M > Th3 then new mode is A; if M ⁇ Th4 then new mode is C;
  • the values of the thresholds Th 1 to Th6 should be chosen such that the overall operation of the system exhibits hysteresis. That it to say, if the values of M for different methods were comparable, then:
  • Th3 71 and Th5 > TV, Th 1 > Th6; Th6 > T2 and Th6 > T4.
  • a speech file of a suitable duration for example 2 minutes, is processed through a simulation of the speech and channel encoders ( 14, 1 6) to produce a reference channel file.
  • This file is then further processed through the simulated speech and channel decoder to produce a reference speech output file.
  • a segmental SNR which is known to the art, is calculated between the reference output file, and a degraded output file that is produced from the reference channel file with the f h bit in class k inverted in every 10
  • Q k is the segmental SNR expressed in dB, calculated between the reference output file and the degraded output file that was produced by inverting the I th bit in class k for every frame.
  • cyclic redundancy check (CRC) bits are employed to detect uncorrected errors in the most sensitive bits.
  • CRC cyclic redundancy check
  • a bad frame is declared and an error concealment algorithm is invoked that estimates the corrupted parameters on the basis of their most recent values.
  • the output gain is usually attenuated over successive bad frames until the signal is completely muted.
  • a typical muting period might be five 20ms frames. Since the error concealment algorithm will generally reduce the subjective effect of bad frames, in the case of detected bad frames, the metric in Equation (5) can be replaced by a pre-determined constant, which can be decreased over successive bad frames to represent the effect of the muting algorithm. Metric values for bad frames can be determined in a similar manner to the values for r k .
  • the measure M is calculated using a so-called 'bottom tracking' algorithm rather than the moving average filter described above.
  • the log weighted sum obtained for the previous frame in step 72 is increased by one eighth of a decibel and compared to the log weighted sum obtained for the current frame. If the increased log weighted sum is less than or equal to the log 1 1
  • the measure is set to the former. If, however, the contrary is true, then the measure is set to the latter.
  • Equation (2) the probability measures for the different bits within a significance class are averaged to provide a single value for each class. In alternative embodiments this averaging step is avoided, the values Pi for each bit being used instead.
  • the mobile telephone apparatus 10 may carry out the calculation of the degradation measure rather than the base station 30.
  • the data stream may, for example, include encoded video, facsimile or computer data or a mixture of these.
  • the transmission media may, for example, be microwaves or electrical signals over copper wires.

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  • Engineering & Computer Science (AREA)
  • Computer Networks & Wireless Communication (AREA)
  • Signal Processing (AREA)
  • Quality & Reliability (AREA)
  • Artificial Intelligence (AREA)
  • Power Engineering (AREA)
  • Detection And Prevention Of Errors In Transmission (AREA)
  • Mobile Radio Communication Systems (AREA)

Abstract

On décrit un procédé qui permet de mesurer la dégradation d'un flux de données provoquée par l'influence des erreurs. A la différence des mesures de dégradation connues, une mesure calculée selon la présente invention prend en compte la variation du sens d'une erreur en fonction des éléments qui sont affectés par cette dernière. On décrit une application particulière de ce procédé pour sélectionner un procédé de codage de la parole parmi un certain nombre de procédés possibles de codage de la parole destiné à être utilisé pour transmettre des signaux vocaux sur une liaison radio (20) entre un téléphone mobile (10) et une station de base (30).
PCT/GB1999/000731 1998-03-12 1999-03-11 Procede et appareil de mesure de la degradation du signal Ceased WO1999046884A1 (fr)

Priority Applications (2)

Application Number Priority Date Filing Date Title
EP99907765A EP1062758A1 (fr) 1998-03-12 1999-03-11 Procede et appareil de mesure de la degradation du signal
US09/622,221 US6714896B1 (en) 1998-03-12 1999-03-11 Method and apparatus for signal degradation measurement

Applications Claiming Priority (4)

Application Number Priority Date Filing Date Title
GB9805330.9 1998-03-12
EP98301853 1998-03-12
EP98301853.2 1998-03-12
GBGB9805330.9A GB9805330D0 (en) 1998-03-12 1998-03-12 Signal degradation measurement

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WO1999046884A1 true WO1999046884A1 (fr) 1999-09-16

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EP (1) EP1062758A1 (fr)
WO (1) WO1999046884A1 (fr)

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US7406539B2 (en) * 2000-10-17 2008-07-29 Avaya Technology Corp. Method and apparatus for performance and cost optimization in an internetwork
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EP1062758A1 (fr) 2000-12-27
US6714896B1 (en) 2004-03-30

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